1 // Copyright 2013 The Chromium Authors. All rights reserved. 2 // Use of this source code is governed by a BSD-style license that can be 3 // found in the LICENSE file. 4 5 #ifndef MEDIA_CAST_RTCP_RTCP_H_ 6 #define MEDIA_CAST_RTCP_RTCP_H_ 7 8 #include <map> 9 #include <queue> 10 #include <string> 11 12 #include "base/basictypes.h" 13 #include "base/memory/scoped_ptr.h" 14 #include "base/time/tick_clock.h" 15 #include "base/time/time.h" 16 #include "media/cast/base/clock_drift_smoother.h" 17 #include "media/cast/cast_config.h" 18 #include "media/cast/cast_defines.h" 19 #include "media/cast/cast_environment.h" 20 #include "media/cast/rtcp/receiver_rtcp_event_subscriber.h" 21 #include "media/cast/rtcp/rtcp_defines.h" 22 #include "media/cast/transport/cast_transport_defines.h" 23 #include "media/cast/transport/cast_transport_sender.h" 24 #include "media/cast/transport/pacing/paced_sender.h" 25 26 namespace media { 27 namespace cast { 28 29 class LocalRtcpReceiverFeedback; 30 class LocalRtcpRttFeedback; 31 class PacedPacketSender; 32 class RtcpReceiver; 33 class RtcpSender; 34 35 typedef std::pair<uint32, base::TimeTicks> RtcpSendTimePair; 36 typedef std::map<uint32, base::TimeTicks> RtcpSendTimeMap; 37 typedef std::queue<RtcpSendTimePair> RtcpSendTimeQueue; 38 39 class RtcpSenderFeedback { 40 public: 41 virtual void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback) = 0; 42 43 virtual ~RtcpSenderFeedback() {} 44 }; 45 46 class RtpReceiverStatistics { 47 public: 48 virtual void GetStatistics(uint8* fraction_lost, 49 uint32* cumulative_lost, // 24 bits valid. 50 uint32* extended_high_sequence_number, 51 uint32* jitter) = 0; 52 53 virtual ~RtpReceiverStatistics() {} 54 }; 55 56 class Rtcp { 57 public: 58 // Rtcp accepts two transports, one to be used by Cast senders 59 // (CastTransportSender) only, and the other (PacedPacketSender) should only 60 // be used by the Cast receivers and test applications. 61 Rtcp(scoped_refptr<CastEnvironment> cast_environment, 62 RtcpSenderFeedback* sender_feedback, 63 transport::CastTransportSender* const transport_sender, // Send-side. 64 transport::PacedPacketSender* paced_packet_sender, // Receive side. 65 RtpReceiverStatistics* rtp_receiver_statistics, 66 RtcpMode rtcp_mode, 67 const base::TimeDelta& rtcp_interval, 68 uint32 local_ssrc, 69 uint32 remote_ssrc, 70 const std::string& c_name, 71 EventMediaType event_media_type); 72 73 virtual ~Rtcp(); 74 75 static bool IsRtcpPacket(const uint8* rtcp_buffer, size_t length); 76 77 static uint32 GetSsrcOfSender(const uint8* rtcp_buffer, size_t length); 78 79 base::TimeTicks TimeToSendNextRtcpReport(); 80 81 // Send a RTCP sender report. 82 // |current_time| is the current time reported by a tick clock. 83 // |current_time_as_rtp_timestamp| is the corresponding RTP timestamp. 84 void SendRtcpFromRtpSender(base::TimeTicks current_time, 85 uint32 current_time_as_rtp_timestamp); 86 87 // |cast_message| and |rtcp_events| is optional; if |cast_message| is 88 // provided the RTCP receiver report will append a Cast message containing 89 // Acks and Nacks; if |rtcp_events| is provided the RTCP receiver report 90 // will append the log messages. 91 void SendRtcpFromRtpReceiver( 92 const RtcpCastMessage* cast_message, 93 const ReceiverRtcpEventSubscriber::RtcpEventMultiMap* rtcp_events); 94 95 void IncomingRtcpPacket(const uint8* rtcp_buffer, size_t length); 96 97 // TODO(miu): Clean up this method and downstream code: Only VideoSender uses 98 // this (for congestion control), and only the |rtt| and |avg_rtt| values, and 99 // it's not clear that any of the downstream code is doing the right thing 100 // with this data. 101 bool Rtt(base::TimeDelta* rtt, 102 base::TimeDelta* avg_rtt, 103 base::TimeDelta* min_rtt, 104 base::TimeDelta* max_rtt) const; 105 106 bool is_rtt_available() const { return number_of_rtt_in_avg_ > 0; } 107 108 // If available, returns true and sets the output arguments to the latest 109 // lip-sync timestamps gleaned from the sender reports. While the sender 110 // provides reference NTP times relative to its own wall clock, the 111 // |reference_time| returned here has been translated to the local 112 // CastEnvironment clock. 113 bool GetLatestLipSyncTimes(uint32* rtp_timestamp, 114 base::TimeTicks* reference_time) const; 115 116 // Set the history size to record Cast receiver events. The event history is 117 // used to remove duplicates. The history will store at most |size| events. 118 void SetCastReceiverEventHistorySize(size_t size); 119 120 // Update the target delay. Will be added to every report sent back to the 121 // sender. 122 // TODO(miu): Remove this deprecated functionality. The sender ignores this. 123 void SetTargetDelay(base::TimeDelta target_delay); 124 125 void OnReceivedReceiverLog(const RtcpReceiverLogMessage& receiver_log); 126 127 protected: 128 void OnReceivedNtp(uint32 ntp_seconds, uint32 ntp_fraction); 129 void OnReceivedLipSyncInfo(uint32 rtp_timestamp, 130 uint32 ntp_seconds, 131 uint32 ntp_fraction); 132 133 private: 134 friend class LocalRtcpRttFeedback; 135 friend class LocalRtcpReceiverFeedback; 136 137 void OnReceivedDelaySinceLastReport(uint32 receivers_ssrc, 138 uint32 last_report, 139 uint32 delay_since_last_report); 140 141 void OnReceivedSendReportRequest(); 142 143 void UpdateRtt(const base::TimeDelta& sender_delay, 144 const base::TimeDelta& receiver_delay); 145 146 void UpdateNextTimeToSendRtcp(); 147 148 void SaveLastSentNtpTime(const base::TimeTicks& now, 149 uint32 last_ntp_seconds, 150 uint32 last_ntp_fraction); 151 152 scoped_refptr<CastEnvironment> cast_environment_; 153 transport::CastTransportSender* const transport_sender_; 154 const base::TimeDelta rtcp_interval_; 155 const RtcpMode rtcp_mode_; 156 const uint32 local_ssrc_; 157 const uint32 remote_ssrc_; 158 const std::string c_name_; 159 const EventMediaType event_media_type_; 160 161 // Not owned by this class. 162 RtpReceiverStatistics* const rtp_receiver_statistics_; 163 164 scoped_ptr<LocalRtcpRttFeedback> rtt_feedback_; 165 scoped_ptr<LocalRtcpReceiverFeedback> receiver_feedback_; 166 scoped_ptr<RtcpSender> rtcp_sender_; 167 scoped_ptr<RtcpReceiver> rtcp_receiver_; 168 169 base::TimeTicks next_time_to_send_rtcp_; 170 RtcpSendTimeMap last_reports_sent_map_; 171 RtcpSendTimeQueue last_reports_sent_queue_; 172 173 // The truncated (i.e., 64-->32-bit) NTP timestamp provided in the last report 174 // from the remote peer, along with the local time at which the report was 175 // received. These values are used for ping-pong'ing NTP timestamps between 176 // the peers so that they can estimate the network's round-trip time. 177 uint32 last_report_truncated_ntp_; 178 base::TimeTicks time_last_report_received_; 179 180 // Maintains a smoothed offset between the local clock and the remote clock. 181 // Calling this member's Current() method is only valid if 182 // |time_last_report_received_| is not "null." 183 ClockDriftSmoother local_clock_ahead_by_; 184 185 // Latest "lip sync" info from the sender. The sender provides the RTP 186 // timestamp of some frame of its choosing and also a corresponding reference 187 // NTP timestamp sampled from a clock common to all media streams. It is 188 // expected that the sender will update this data regularly and in a timely 189 // manner (e.g., about once per second). 190 uint32 lip_sync_rtp_timestamp_; 191 uint64 lip_sync_ntp_timestamp_; 192 193 base::TimeDelta rtt_; 194 base::TimeDelta min_rtt_; 195 base::TimeDelta max_rtt_; 196 int number_of_rtt_in_avg_; 197 double avg_rtt_ms_; 198 uint16 target_delay_ms_; 199 200 DISALLOW_COPY_AND_ASSIGN(Rtcp); 201 }; 202 203 } // namespace cast 204 } // namespace media 205 206 #endif // MEDIA_CAST_RTCP_RTCP_H_ 207
